[Asterisk-Users] testing asterisk on FXS lines
Jason Kawakami
jkkawakami at optellabs.com
Mon May 24 17:46:22 MST 2004
i always use the Goto application. seems to work quite well for testing
those "s" extensions.
exten = 2500,1,Goto(context,s,1)
will take you to step 1 in the s extension in whatever context.
Jason Kawakami
----- Original Message -----
From: <asterisk-users-request at lists.digium.com>
To: <asterisk-users at lists.digium.com>
Sent: Monday, May 24, 2004 5:20 PM
Subject: Asterisk-Users digest, Vol 1 #3886 - 9 msgs
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> Today's Topics:
>
> 1. Re: Re: Making a SIP call (Eric Wieling)
> 2. RE: testing asterisk on FXS lines (Jay Milk)
> 3. SIP Authentication Problem (Chuck Ramirez)
> 4. RE: 2 Sip phones behind un-natted Asterisk (Chad Brown)
> 5. Re: extensions/sip from database? (Fran Boon)
> 6. Using Blacklist (Steven E. Frazier)
> 7. Asterisk connected to DataBase (pesb)
> 8. mpg123 (Simon Brown)
> 9. Re: Using Blacklist (Dorian Gray)
>
> --__--__--
>
> Message: 1
> Date: Mon, 24 May 2004 16:20:36 -0500
> From: Eric Wieling <eric at fnords.org>
> To: asterisk-users at lists.digium.com
> Subject: Re: [Asterisk-Users] Re: Making a SIP call
> Reply-To: asterisk-users at lists.digium.com
>
> bclark at bwkip.com wrote:
> > I am still having this problem of only capturing part of the IP address,
I
> > am currently checking into a possible hardware/software issue on the
> > client side but was wondering if there are any setting I need to set on
> > the asterisk server to allow an peer to peer call. I have set
> > dtmfmode=inband. Is there anything else I need to set?
>
> dtmfmode=inband only works with the ulaw and alaw codecs. If you use
> any other codec you MUST use rfc2833 or info DTMF modes (set on the
> phone AND on Asterisk)
>
> --__--__--
>
> Message: 2
> From: "Jay Milk" <jay at skimmilk.net>
> To: <asterisk-users at lists.digium.com>
> Subject: RE: [Asterisk-Users] testing asterisk on FXS lines
> Date: Mon, 24 May 2004 16:29:39 -0500
> Reply-To: asterisk-users at lists.digium.com
>
> For $49.99+S&H I can sell you an FXO/FXS test-cable... just kidding.
> Use a regular RJ11 cable to connect one of your FXS ports to the FXO
> port you want to test, pick up another FXS and dial the extension... and
> you're promptly delivered to the [incoming] context. I test all my FXO
> configs using a Sipura FXS port to make it ring. I'd still like that
> $50 though :)
>
> -----Original Message-----
> From: asterisk-users-admin at lists.digium.com
> [mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Michael
> George
> Sent: Monday, May 24, 2004 3:57 PM
> To: asterisk-users at lists.digium.com
> Subject: Re: [Asterisk-Users] testing asterisk on FXS lines
>
>
> On May 24, 2004, at 4:00 PM, Michael George wrote:
> > I am configuring an asterisk server and I want to test the incoming
> > configuration with my FXS handsets.
> >
> > I have the FXS lines able to call eachother and they can connect out
> > the FXO lines.
> >
> > I changed the context for the FXS lines to "incoming" so that they
> > would be able to test the setup for incoming calls.
> >
> > For the incoming context I have:
> > [incoming]
> > exten => s,1,Wait(1)
> > exten => s,2,Answer()
> > exten => s,3,Background(hello2) ; this is the file I need to test the
> > playback of first
> >
> > And I do a restart. When I pickup one of the FXS handsets, though, I
> > get this from asterisk (running with the -vvvc arg):
> > Starting simple switch on 'Zap/1-1'
> > and that is it.
> >
> > I know that the context is right because I put a hard-dial of "202" in
> > there and when I dialed it, it would connect to that extension (Zap/2)
>
> > and if I dialed anything else I would get fast busy.
> >
> > I have checked and the line right after the last exten above is
> > another context marker.
> >
> > The asterisk output also shows the s extensions being loaded under the
> > correct context when I do a reload after the restart (to see just the
> > messages from the contexts being loaded).
> >
> > What am I missing to get the FXS lines, in the context "incoming", to
> > do the wait/answer/background?
> >
> > Thanks!
>
> For some reason, the s extension is not being executed for the FXS
> lines. I changed their default context back to "internal" and added
> "exten => s,1,Background(hello2)" to the internal context, thinking
> that when I pick up the handset I will get the hello2 audio file played
> as it waits for me to enter digits.
>
> But the audio file is not played... I must be missing an essential
> concept here...
>
> -Michael
>
> _______________________________________________
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> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
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>
>
> --__--__--
>
> Message: 3
> Date: Mon, 24 May 2004 14:25:00 -0700 (PDT)
> From: Chuck Ramirez <chuck_ramirez at yahoo.com>
> To: asterisk-users at lists.digium.com
> Subject: [Asterisk-Users] SIP Authentication Problem
> Reply-To: asterisk-users at lists.digium.com
>
> --0-909188567-1085433900=:35567
> Content-Type: text/plain; charset=us-ascii
>
>
> I have a group of users configured as extensions in *.These users are
registered with a SIP Proxy Server and can receive calls very well. The
problem happens when any user tries to make an outbound call. The proxy
replies with a "401 Unauthorized" and * don't try another INVITE including
credentials.
>
> Here is part of the content of sip.conf.
>
> [general]
> port = 5061
> bindaddr = *.IP
> context = invalidcalls
>
> ;This account is used for inbound and outbound calls
> register => myuser:mypass at mydomain/999
>
> [mydomain]
> type=peer
> host=myproxy
> context=sip
> username=myuser
> secret=mypass
> fromuser=myuser
> fromdomain=mydomain
>
> [user1]
> type=friend
> host=dynamic
> defaultip=default.IP
> username=user1
> secret=secret1
> dtmfmode=rfc2833
> context=users
> callerid="User 1"
> nat=yes
>
>
>
> Here is part of the content of extensions.conf.
>
> ;This part is working fine
> [sip]
> exten => 999,1,Dial(SIP/user1,,tr)
>
> [users]
> exten => _8.,1,Dial,SIP/${EXTEN-1}@mydomain,tr
>
>
>
> When I dial the number 812345 from my SIP Phone, this is the message
sequence
> Phone -> Asterisk: INVITE sip:812345@*domain SIP/2.0
> Asterisk -> Phone: SIP/2.0 407 Proxy Authentication Required
> Phone -> Asterisk: ACK sip:812345@*domain SIP/2.0
> Phone -> Asterisk: INVITE sip:812345@*domain SIP/2.0 (with authentication
header)
> Asterisk -> Phone: SIP/2.0 100 Trying
> Asterisk -> Proxy: INVITE sip:12345 at mydomain SIP/2.0
> Proxy -> Asterisk: SIP/2.0 401 Unauthorized
> Asterisk -> Proxy: ACK sip:12345 at mydomain SIP/2.0
>
> The next message I would expect is another INVITE from * to the proxy with
the authentication header.
> Why * hasn't send it? Can someone give me a help?
>
> Thanks in advance
> Chuck Ramirez
>
>
>
>
> ---------------------------------
> Do you Yahoo!?
> Friends. Fun. Try the all-new Yahoo! Messenger
> --0-909188567-1085433900=:35567
> Content-Type: text/html; charset=us-ascii
>
> <P align=left>I have a group of users configured as extensions in *.These
users are registered with a SIP Proxy Server and can receive calls very
well. The problem happens when any user tries to make an outbound call. The
proxy replies with a "401 Unauthorized" and * don't try another INVITE
including credentials.</P>
> <P align=left>Here is part of the content of sip.conf.</P>
> <P align=left>[general]<BR>port = 5061<BR>bindaddr = *.IP<BR>context =
invalidcalls</P>
> <P align=left>;This account is used for inbound and outbound
calls<BR>register => myuser:mypass at mydomain/999</P>
> <P
align=left>[mydomain]<BR>type=peer<BR>host=myproxy<BR>context=sip<BR>usernam
e=myuser<BR>secret=mypass<BR>fromuser=myuser<BR>fromdomain=mydomain</P>
> <P
align=left>[user1]<BR>type=friend<BR>host=dynamic<BR>defaultip=default.IP<BR
>username=user1<BR>secret=secret1<BR>dtmfmode=rfc2833<BR>context=users<BR>ca
llerid="User 1"<BR>nat=yes</P>
> <P align=left> </P>
> <P align=left>Here is part of the content of extensions.conf.</P>
> <P align=left>;This part is working fine<BR>[sip]<BR>exten =>
999,1,Dial(SIP/user1,,tr)</P>
> <P align=left>[users]<BR>exten =>
_8.,1,Dial,SIP/${EXTEN-1}@mydomain,tr</P>
> <P align=left> </P>
> <P align=left>When I dial the number 812345 from my SIP Phone, this is the
message sequence<BR>Phone -> Asterisk: INVITE sip:812345@*domain
SIP/2.0<BR>Asterisk -> Phone: SIP/2.0 407 Proxy Authentication
Required<BR>Phone -> Asterisk: ACK sip:812345@*domain
SIP/2.0<BR>Phone -> Asterisk: INVITE sip:812345@*domain SIP/2.0 (with
authentication header)<BR>Asterisk -> Phone: SIP/2.0 100
Trying<BR>Asterisk -> Proxy: INVITE sip:12345 at mydomain
SIP/2.0<BR>Proxy -> Asterisk: SIP/2.0 401 Unauthorized<BR>Asterisk ->
Proxy: ACK sip:12345 at mydomain SIP/2.0</P>
> <P align=left>The next message I would expect is another INVITE from * to
the proxy with the authentication header.<BR>Why * hasn't send it? Can
someone give me a help?</P>
> <P align=left>Thanks in advance<BR> Chuck
Ramirez</P><BR><BR><p>
> <hr size=1><font face=arial size=-1>Do you Yahoo!?<br>Friends. Fun. <a
href="http://messenger.yahoo.com/">Try the all-new Yahoo! Messenger</a>
> --0-909188567-1085433900=:35567--
>
> --__--__--
>
> Message: 4
> Subject: RE: [Asterisk-Users] 2 Sip phones behind un-natted Asterisk
> Date: Mon, 24 May 2004 14:36:00 -0700
> From: "Chad Brown" <chad.brown at identitymine.com>
> To: <asterisk-users at lists.digium.com>
> Reply-To: asterisk-users at lists.digium.com
>
> After further investigation it looks like it was as simple as both
> phones trying to listen on the same port. I will continue testing to
> verify.
>
> -----Original Message-----
> From: asterisk-users-admin at lists.digium.com
> [mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Shaun Dawson
> Sent: Monday, May 24, 2004 10:03 AM
> To: asterisk-users at lists.digium.com
> Subject: Re: [Asterisk-Users] 2 Sip phones behind un-natted Asterisk
>
> What does the Xten diagnostic log say about a single
> session?
>
> Also, what does the * SIP debug output say? I'd be
> very interested to see what IPs and ports SIP is
> trying to set the RTP connection on. (Since SIP
> appears to be working fine, it's the RTP part that is
> breaking).
>
> Are both the Xten and the 7960 trying to listen on the
> same RTP port (my Xten is configured to listen on
> 8000)?
>
> Pardon me if I sound like an idiot, but I'm somewhat
> new to VoIP, SIP _and_ Asterisk. :)
>
> Shaun
>
>
> --- Bruce Komito <brucek at bagel.com> wrote:
> > John, In my case, the two ports are not using the
> > same IP port (one is on
> > 5060, the other on 5061), but of course, they are on
> > the same IP address.
> > I think that is what is confusing the NAT server,
> > but I don't know what to
> > do about it.
> >=20
> > Bruce Komito
> > High Sierra Networks, Inc.
> > www.servers-r-us.com
> > (775) 284-5800 ext 115
> >=20
> >=20
> > On Mon, 24 May 2004, John Fraizer wrote:
> >=20
> > > Chad Brown wrote:
> > >
> > > > I have 2 SIP phones (Cisco 7960 & XTen) behind a
> > NAT provided by a
> > > > Linksys firewall that supports UPnP. The
> > Asterisk server has a public
> > > > IP. Here are the problems that I am having with
> > this configuration...
> > > >
> > > >
> > > >
> > > > 1. The 2 SIP phones can call MeetMe and have
> > a conference but cannot
> > > > call each other. (Yes, they connect but no
> > audio either direction)
> > > > 2. I have verify=3Dyes in the sip.conf for both
> > phones. Both phones
> > > > constantly go Unreachable. (However, the
> > connection is very fast
> > > > between * and sip phones)
> > > > 3. Sometimes but not always when I try to
> > call phone1 phone2 rings.
> > > >
> > > >
> > > >
> > > > Is this Nat messing with me or something else?
> > In any case...Any advice
> > > > out there?
> > > >
> > > >
> > > >
> > > > Thanks,
> > > >
> > > > Chad
> > > >
> > >
> > >
> > > The problem is probably that both of your SIP
> > phones are using the same
> > > port. I played with two 7960's behind a Linksys
> > on Saturday and finally
> > > got them playing right when I changed the
> > following:
> > >
> > > In Phone 1's SIP[macaddr].cnf:
> > >
> > > voip_control_port: 5061
> > >
> > > In Phone 2's SIP[macaddr].cnf:
> > >
> > > voip_control_port: 5062
> > >
> > > The default control port is 5060. Note: This is
> > the port that the
> > > PHONE uses to initiate the connection to * and not
> > the port it is
> > > connecting to.
> > >
> > > John
> > > _______________________________________________
> > > Asterisk-Users mailing list
> > > Asterisk-Users at lists.digium.com
> > >
> >
> http://lists.digium.com/mailman/listinfo/asterisk-users
> > > To UNSUBSCRIBE or update options visit:
> > > =20
> >
> http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> >=20
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com
> >
> http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> > =20
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
>
> =09
> =09
> __________________________________
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> Yahoo! Domains - Claim yours for only $14.70/year
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> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
> --__--__--
>
> Message: 5
> Date: Mon, 24 May 2004 22:50:44 +0100
> From: Fran Boon <flavour at partyvibe.com>
> To: asterisk-users at lists.digium.com
> Subject: Re: [Asterisk-Users] extensions/sip from database?
> Reply-To: asterisk-users at lists.digium.com
>
> Manuel Wenger wrote:
> > We are planning to deploy a pretty large asterisk server with many SIP
extensions (might be up to 10000 in the future), and I have a few questions:
> > 1) is this possible, or are we running into some kind of limitation in
the software that I wasn't aware of and that I didn't find by browsing
through the archives and through Wiki? No, we don't need any G729-G711
transformations, it would only be acting as a SIP proxy (even if asterisk
isn't a proxy).
>
> /Should/ be psosible with canreinvite=yes & no use of T,t in the dial
> commands, so that Asterisk can stay out of the media path except when
> absolutely necessary.
>
> > 2) is there a way to store extensions.conf and/or sip.conf in some kind
of database, maybe MySQL? This would make life easier if someone wanted to
change his SIP password. Or how would you otherwise solve this problem?
>
> http://voip-info.org/wiki-Asterisk+configuration+from+database
> Option 1 is being enhanced through the development of ast_data.
> I currently use Option 2
>
> > 3) is there a quick way of reloading only a part of
sip.conf/extensions.conf, for example if only a user password changed, or an
extension's behaviour (eg. routing to voicemail instead of a SIP user)?
>
> sip reload
> extensions reload
>
> That's as granular as it gets.
> Should be harmless to keep doing this, though.
>
> > Maybe I'm looking at the wrong software here and SER would be better for
what I want to do... I know asterisk is supposed to be a PBX replacement,
but the functions and flexibility it has really tells me "stick with
asterisk". Or am I way off with these assumptions?
>
> Possibly - depends whether you're after a SIP proxy or a PBX ;)
>
> F
>
> --__--__--
>
> Message: 6
> From: "Steven E. Frazier" <sfrazier at fraziercorp.com>
> To: <asterisk-users at lists.digium.com>
> Date: Mon, 24 May 2004 17:55:17 -0400
> Subject: [Asterisk-Users] Using Blacklist
> Reply-To: asterisk-users at lists.digium.com
>
> I am attempting to write in incoming context for calls.
>
> 1. If the caller id is given and it is not black listed it will Playback =
> a
> greeting and then right the phone or go to voicemail under busy or
> unavailable conditions
> 2. If no caller id is given, then Privacy Manager will ask for the =
> number.
> I am testing 6145551212 to see if the black list will work
> 3. If a caller id is given, and it is blacklisted (in the blacklist db) =
> I
> would like for it to go to Playback/black-list-blocked message
>
>
>
>
> The db shows:
>
> asterisk*CLI> database show blacklist
> /blacklist/<1010987/18887975686number> : 1
>
> /blacklist/<name/number> : 1
>
> /blacklist/unlisted/6145551212 : 1
>
> asterisk*CLI>
>
>
> exten =3D> 2129,1,Wait(1)
> exten =3D> 2129,2,Zapateller(answer|nocallerid)
> exten =3D> 2129,3,NoOp
> exten =3D> 2129,4,PrivacyManager
> exten =3D> 2129,5,LookupBlacklist
> exten =3D> 2129,6,Dial(Zap/4,5,Ttr)
> exten =3D> 2129,7,Answer
> exten =3D> 2129,8,Wait(1)
> exten =3D> 2129,9,Playback(personal/hello)
> exten =3D> 2129,10,Playback(personal/i-am-not-in-at-the-moment)
> exten =3D> 2129,11,VoiceMail2(u${EXTEN})
> exten =3D> 2129,12,Hangup
> exten =3D> 2129,102,VoiceMail2(b${EXTEN}) ; Busy Voicemail if extension =
> is
> busy
> exten =3D> 2129,106,Playback,personal/black-list-blocked
> exten =3D> 2129,108,Wait(2)
> exten =3D> 2129,110,Hangup
>
> When I dial my test extension of 2129, I get:
>
>
> asterisk*CLI>=20
> -- Starting simple switch on 'Zap/7-1'
> -- Disabling Caller*ID on Zap/7-1
> -- Executing Wait("Zap/7-1", "1") in new stack
> -- Executing Zapateller("Zap/7-1", "answer|nocallerid") in new stack
> -- Executing NoOp("Zap/7-1", "") in new stack
> -- Executing PrivacyManager("Zap/7-1", "") in new stack
> =3D=3D Parsing '/etc/asterisk/privacy.conf': =3D=3D Parsing
> '/etc/asterisk/privacy.conf': Found
> -- Playing 'privacy-unident' (language 'en')
> -- Playing 'privacy-prompt' (language 'en')
> -- Playing 'privacy-thankyou' (language 'en')
> -- Changed Caller*ID to "Privacy Manager" <6145551212>
> -- Executing LookupBlacklist("Zap/7-1", "") in new stack
> -- Executing Dial("Zap/7-1", "Zap/4|5|Ttr") in new stack
> -- Called 4
> -- Zap/4-1 is ringing
> -- Zap/4-1 is ringing
> -- Nobody picked up in 5000 ms
> -- Hungup 'Zap/4-1'
> -- Executing Answer("Zap/7-1", "") in new stack
> -- Executing Wait("Zap/7-1", "1") in new stack
> -- Executing Playback("Zap/7-1", "personal/hello") in new stack
> -- Playing 'personal/hello' (language 'en')
> -- Executing Playback("Zap/7-1", =
> "personal/i-am-not-in-at-the-moment")
> in new stack
> -- Playing 'personal/i-am-not-in-at-the-moment' (language 'en')
> -- Executing VoiceMail2("Zap/7-1", "u2129") in new stack
> -- Playing 'vm-theperson' (language 'en')
> -- Playing 'digits/2' (language 'en')
> -- Playing 'digits/1' (language 'en')
> -- Playing 'digits/2' (language 'en')
> -- Playing 'digits/9' (language 'en')
> -- Playing 'vm-isunavail' (language 'en')
> -- Playing 'vm-intro' (language 'en')
> -- Playing 'beep' (language 'en')
> -- Recording the message
>
> It goes to the unavailable voice mail box.
>
> According to the documentation and my understanding:
>
>
> LookupBlacklist: Looks up the Caller*ID number on the active channel in =
> the
> Asterisk database (family 'blacklist'). If the number is found, and if =
> there
> exists a priority n + 101, where 'n' is the priority of the current
> instance, then the channel will be setup to continue at that priority =
> level.
> Otherwise, it returns 0. Does nothing if no Caller*ID was received on =
> the
> channel.=20
> Example: database put blacklist <name/number> 1
>
>
> Could someone tell me what I am doing wrong that it won't go to Priority =
> 106
> and Playback black-list-blocked.
>
> Would someone share their context that is using blacklist to show me how
> they are doing it?
>
> Thanks.
>
> --__--__--
>
> Message: 7
> From: pesb <pesb at conexion.com.py>
> To: asterisk-users at lists.digium.com
> Date: Mon, 24 May 2004 17:58:30 -0400
> Subject: [Asterisk-Users] Asterisk connected to DataBase
> Reply-To: asterisk-users at lists.digium.com
>
> Hi there,
> I want to have all my sip.conf data inside a DataBase, so that my
asterisk=
> =20
> admintration system would be through a Web Interface connected to the DB.
> Is there any way to put the sip.conf file in a Data Base and then to
read=
> =20
> from it, in such a way that the sip.conf file would have some line that=20
> points to the DataBase?
> I have seen wiki's page=20
> http://www.voip-info.org/wiki-Asterisk+configuration+from+database
> Possibility n=BA 2 and 3 do not convince myself.
> I have tried possibility n=BA1(Dynamic), but did not find much info about
t=
> he=20
> command DBget. Could somebody give some info on how to use it?
> Or, could someone recommend me another scheme that could work?
>
> thanks in advance,
> Pablo Salinas
>
>
>
>
> --__--__--
>
> Message: 8
> Date: Tue, 25 May 2004 08:11:30 +1000
> From: "Simon Brown" <Simon.Brown at otterson.com.au>
> To: <asterisk-users at lists.digium.com>
> Subject: [Asterisk-Users] mpg123
> Reply-To: asterisk-users at lists.digium.com
>
> When I start * I get 6 mpg123 processes start as well. Is this normal?
> Often after a couple of days these mpg123 processes start to consume cpu =
> and
> I have to kill them off.
> I do not have a sound card in the server and I have noload =3D> =
> chan_oss.so
>
> Simon
>
> --__--__--
>
> Message: 9
> Date: Mon, 24 May 2004 18:17:09 -0400
> From: Dorian Gray <asterisk at tintar.com>
> To: asterisk-users at lists.digium.com
> Subject: Re: [Asterisk-Users] Using Blacklist
> Reply-To: asterisk-users at lists.digium.com
>
> the following has been working well for me, and I think it does similar
> to what you want...
>
>
> [macro-blackdrop]
> exten => s,1,Playback(giggle1)
> ; something is terribly wrong...!
> exten => s,2,Playback(tt-somethingwrong)
> ; oh, it's those damnable weasels again...!
> exten => s,3,Playback(tt-weasels)
> exten => s,4,Playback(goodbye)
> exten => s,5,Hangup
>
> [inbound-analog]
> exten => s,1,SetMusicOnHold,random
> exten => s,2,Zapateller(answer|nocallerid)
> exten => s,3,NoOp
> exten => s,4,PrivacyManager
> exten => s,5,LookupCIDName
> exten => s,6,LookupBlacklist
> exten => s,7,Background(pls-wait-connect-call)
> exten => s,8,Dial(${PHONE1}&${PHONES1},20,Ttm)
> exten => s,9,Answer
> exten => s,10,Wait(1)
> exten => s,11,Macro(vmessage,${PHONE1VM})
> exten => s,105,Macro(blackdrop)
> exten => s,107,Macro(blackdrop)
>
> hm maybe I should move lookupcidname after lookupblacklist and save a
> few cycles ^_^
>
>
>
> Steven E. Frazier wrote:
> > I am attempting to write in incoming context for calls.
> >
> > 1. If the caller id is given and it is not black listed it will Playback
a
> > greeting and then right the phone or go to voicemail under busy or
> > unavailable conditions
> > 2. If no caller id is given, then Privacy Manager will ask for the
number.
> > I am testing 6145551212 to see if the black list will work
> > 3. If a caller id is given, and it is blacklisted (in the blacklist db)
I
> > would like for it to go to Playback/black-list-blocked message
> >
> >
> >
> >
> > The db shows:
> >
> > asterisk*CLI> database show blacklist
> > /blacklist/<1010987/18887975686number> : 1
> >
> > /blacklist/<name/number> : 1
> >
> > /blacklist/unlisted/6145551212 : 1
> >
> > asterisk*CLI>
> >
> >
> > exten => 2129,1,Wait(1)
> > exten => 2129,2,Zapateller(answer|nocallerid)
> > exten => 2129,3,NoOp
> > exten => 2129,4,PrivacyManager
> > exten => 2129,5,LookupBlacklist
> > exten => 2129,6,Dial(Zap/4,5,Ttr)
> > exten => 2129,7,Answer
> > exten => 2129,8,Wait(1)
> > exten => 2129,9,Playback(personal/hello)
> > exten => 2129,10,Playback(personal/i-am-not-in-at-the-moment)
> > exten => 2129,11,VoiceMail2(u${EXTEN})
> > exten => 2129,12,Hangup
> > exten => 2129,102,VoiceMail2(b${EXTEN}) ; Busy Voicemail if extension is
> > busy
> > exten => 2129,106,Playback,personal/black-list-blocked
> > exten => 2129,108,Wait(2)
> > exten => 2129,110,Hangup
> >
> > When I dial my test extension of 2129, I get:
> >
> >
> > asterisk*CLI>
> > -- Starting simple switch on 'Zap/7-1'
> > -- Disabling Caller*ID on Zap/7-1
> > -- Executing Wait("Zap/7-1", "1") in new stack
> > -- Executing Zapateller("Zap/7-1", "answer|nocallerid") in new stack
> > -- Executing NoOp("Zap/7-1", "") in new stack
> > -- Executing PrivacyManager("Zap/7-1", "") in new stack
> > == Parsing '/etc/asterisk/privacy.conf': == Parsing
> > '/etc/asterisk/privacy.conf': Found
> > -- Playing 'privacy-unident' (language 'en')
> > -- Playing 'privacy-prompt' (language 'en')
> > -- Playing 'privacy-thankyou' (language 'en')
> > -- Changed Caller*ID to "Privacy Manager" <6145551212>
> > -- Executing LookupBlacklist("Zap/7-1", "") in new stack
> > -- Executing Dial("Zap/7-1", "Zap/4|5|Ttr") in new stack
> > -- Called 4
> > -- Zap/4-1 is ringing
> > -- Zap/4-1 is ringing
> > -- Nobody picked up in 5000 ms
> > -- Hungup 'Zap/4-1'
> > -- Executing Answer("Zap/7-1", "") in new stack
> > -- Executing Wait("Zap/7-1", "1") in new stack
> > -- Executing Playback("Zap/7-1", "personal/hello") in new stack
> > -- Playing 'personal/hello' (language 'en')
> > -- Executing Playback("Zap/7-1",
"personal/i-am-not-in-at-the-moment")
> > in new stack
> > -- Playing 'personal/i-am-not-in-at-the-moment' (language 'en')
> > -- Executing VoiceMail2("Zap/7-1", "u2129") in new stack
> > -- Playing 'vm-theperson' (language 'en')
> > -- Playing 'digits/2' (language 'en')
> > -- Playing 'digits/1' (language 'en')
> > -- Playing 'digits/2' (language 'en')
> > -- Playing 'digits/9' (language 'en')
> > -- Playing 'vm-isunavail' (language 'en')
> > -- Playing 'vm-intro' (language 'en')
> > -- Playing 'beep' (language 'en')
> > -- Recording the message
> >
> > It goes to the unavailable voice mail box.
> >
> > According to the documentation and my understanding:
> >
> >
> > LookupBlacklist: Looks up the Caller*ID number on the active channel in
the
> > Asterisk database (family 'blacklist'). If the number is found, and if
there
> > exists a priority n + 101, where 'n' is the priority of the current
> > instance, then the channel will be setup to continue at that priority
level.
> > Otherwise, it returns 0. Does nothing if no Caller*ID was received on
the
> > channel.
> > Example: database put blacklist <name/number> 1
> >
> >
> > Could someone tell me what I am doing wrong that it won't go to Priority
106
> > and Playback black-list-blocked.
> >
> > Would someone share their context that is using blacklist to show me how
> > they are doing it?
> >
> > Thanks.
> > _______________________________________________
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>
>
>
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